EP1310943B1 - Procédé et dispositif de codage et de décodage de signaux de parole - Google Patents

Procédé et dispositif de codage et de décodage de signaux de parole Download PDF

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Publication number
EP1310943B1
EP1310943B1 EP02025094A EP02025094A EP1310943B1 EP 1310943 B1 EP1310943 B1 EP 1310943B1 EP 02025094 A EP02025094 A EP 02025094A EP 02025094 A EP02025094 A EP 02025094A EP 1310943 B1 EP1310943 B1 EP 1310943B1
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Prior art keywords
section
bits
assigned
speech
optimal value
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German (de)
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EP1310943A2 (fr
EP1310943A3 (fr
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Yutaka Banba
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Panasonic Holdings Corp
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Matsushita Electric Industrial Co Ltd
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/04—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using predictive techniques
    • G10L19/08—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters
    • G10L19/10—Determination or coding of the excitation function; Determination or coding of the long-term prediction parameters the excitation function being a multipulse excitation
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
    • G10L19/0208—Subband vocoders

Definitions

  • the present invention relates to a speech coding apparatus, speech decoding apparatus and speech coding/decoding method in sub-band ADPCM (Adaptive Differential Pulse Code Modulation).
  • ADPCM Adaptive Differential Pulse Code Modulation
  • FIG.1 is a block diagram illustrating configurations of speech coding apparatus 300 and speech decoding apparatus 400 used in two-sub-band ADPCM described in Recommendation G.722.
  • Speech coding apparatus 300 is comprised of 24-tap splitting filter bank 310 that splits a frequency band of an input signal to two sub-bands and outputs sub-band signals, ADPCM quantizers 320a and 320b that quantize respective two-split-sub-band signals, and multiplexer 330 that multiplexes codewords quantized in ADPCM quantizers 320a and 320b to produce a bit stream.
  • speech decoding apparatus 400 is comprised of demultiplexer 410 that outputs codewords for each sub-band obtained from transmitted data streams, ADPCM dequantizers 420a and 420b that dequnantize respective codewords for each sub-band output from demuletiplexer 410 to output sub-band signals, and 24-tap synthesis filter bank 430 that performs synthesis filtering on the sub-band signals.
  • a frequency band of an input signal is split to two sub-bands in splitting filter bank 310 and two sub-band signals are generated.
  • Each of the sub-band signals is assigned a predetermined number of quantizing bits and quantized in respective one of ADPCM quantizers 320a and 320b.
  • the codewords obtained by quantization are multiplexed in multiplexer 330 to be bit streams.
  • the bit streams with a plurality of multiplexed codewords are demulitiplexed in demultiplexer 410 to be codewords for each sub-band.
  • the codewords for each sub-band obtained by demultiplexing are dequantized in ADPCM dequantizers 420a and 420b to be sub-band signals.
  • the sub-band signals are subjected to synthesis in synthesis filter bank 430 to be a decoded signal.
  • US Patent 5,974,380 discloses a speech coding/decoding apparatus and method according to the preamble of claims 1, 8 and 16. It employs perfect/non-perfect reconstruction filters, predictive/non-predictive subband encoding, transient analysis, and psychoacoustic/minimum mean square error bit allocation over time, frequency and the multiple audio channels to encode/decode a data stream to generate high fidelity reconstructed audio.
  • the bit allocation for the ADPCM quantizer is determined for each subframe per subband audio channel based on the subband signal input to the subband encoder. Bit allocation indexes are transmitted to the decoder either using integer code or an entropy table.
  • FIG.2 is a block diagram illustrating a configuration of a speech coding apparatus according to the first embodiment of the present invention.
  • splitting filter bank 100 splits a frequency band of an input signal into four sub-bands with the same bandwidth, and performs thinning processing using "4" that is the number of splits, as a thinning number.
  • Band splitting FIR filters 110a to 110d in splitting filter bank 100 perform splitting filtering on an input signal for predetermined frequency bands.
  • Splitting filter bank 100 is a cosine modulation filter bank, and impulse responses of band splitting FIR filters 110a to 110d that are basic filters are asymmetric.
  • downsamplers 120a to 120d in splitting filter bank 100 perform the thinning processing on respective outputs of band splitting FIR filters 110a to 110d for coding efficiency, using, as the number of thinning, "4" equal to the number of splits in splitting filter bank 100, and output respective sub-band signals.
  • Each of ADPCM quantizers 130a to 130d quantizes a residual signal between the respective sub-band signal and a prediction value calculated from the last frame of the sub-band signal to output a scalable codeword. Further, each of ADPCM quantizers 130a to 130d calculates a dequantized value and scale factor from the residual signal.
  • Adaptive bit assigner 140 determines the number of quantizing bits to assign to each of residual signals based on an energy value of the dequantized value calculated in respective one of ADPCM quantizers 130a to 130d.
  • Multiplexer 150 multiplexes codewords output from ADPCM quantizers 130a to 130d to produce a bit stream that is a multiplexed signal.
  • FIG.3 is a block diagram illustrating a primary configuration of the speech coding apparatus according to the first embodiment of the present invention. While FIG.3 illustrates a configuration of ADPCM quantizer 130a and adaptive bit assigner 140, the other ADPCM quantizers, 130b to 130d, have the same configuration as that of the quantizer 130a, and are connected to adaptive bit assigner 140.
  • adder 131 calculates a difference between the sub-band signal input to respective one of ADPCM quantizers 130a to 130d and a prediction value to generate a residual signal.
  • Quantizing section 132 quantizes the generated residual signal using the scale factor, and outputs a codeword with the number of quantizing bits determined in adaptive bit assigner 140.
  • Core bit extracting section 133 deletes least significant bits (hereinafter, referred to as "LSB") from the codeword output from quantizing section 132 to extract core bits.
  • Scale factor adapting section 134 calculates a scale factor from the extracted core bits.
  • Dequantizing section 135 dequantizes the extracted core bits, and outputs a dequantized value to predicting section 136, adder 137, and adaptive bit assigner 140.
  • Predicting section 136 performs zero prediction and pole prediction using the dequantized value and an output of the predicting section 136, and calculates a prediction value of a next frame of the sub-band signal.
  • Adder 137 calculates the sum of the dequantized value and the prediction value calculated in predicting section 136.
  • a speech signal input to the speech coding apparatus is split into four sub-band signals in splitting filter bank 100. Since splitting filter bank 100 is a cosine modulation filter bank and impulse responses of band splitting FIR filters 110a to 110d that are basic filters are asymmetric, a group delay occurring in filtering is decreased, and it is thereby possible to reduce an amount of computation.
  • the split sub-band signals are input to ADPCM quantizers 130a to 130d respectively.
  • Adder 131 calculates a residual signal between the sub-band signal input to respective one of ADPCM quantizers 130a to 130d and a prediction value calculated from the last frame in predicting section 136, and inputs the calculated residual signal to quantizing section 132 .
  • the residual signal is quantized in quantizing section 132 to be a codeword with the number of quantizing bits assigned by adaptive bit assigner 140.
  • Quantizing the residual signal uses the scale factor calculated in scale factor adapting section 134.
  • the codeword quantized in quantizing section 132 is output to multiplexer 150, and also to core bit extracting section 133.
  • the section 133 deletes LSB to extract core bits.
  • the extracted core bits are input to scale factor adapting section 134 to be used in calculating a scale factor, and also to dequantizing section 135.
  • the codeword quantized in quantizing section 132 becomes scalable to keep the consistency of the scale factor.
  • Dequantizing section 135 dequantizes the core bits using the scale factor calculated in scale factor adapting section 134.
  • the dequantized value obtained by dequantizing the core bits is input to predicting section 136.
  • This input value is called a zero prediction input value.
  • the dequantized value is added in adder 137 to a prediction value of a last frame output from predicting section 136, and is input again to predicting section 136.
  • This input value is called a pole prediction input value.
  • predicting section 136 calculates a prediction value of a next frame of the sub-band signal.
  • the dequantized value is input to adaptive bit assigner 140 per a predetermined number of frames such as a pitch period basis.
  • Adaptive bit assigner 140 calculates an energy of the dequantized value, i.e., square sum of the dequantized value as a sample, output from each of ADPCM quantizers 130a to 130d, and based on the calculated energy of the dequantized value, determines the number of bits assigned to each residual signal to be quantized in respective one of ADPCM quantizers 130a to 130d.
  • the determined numbers of quantizing bits are output to respective quantizing sections 132 in ADPCM quantizers 130a to 130d. As described above, each quantizing section 132 quantizes the residual signal of the next frame using the scale factor, and outputs a codeword with the number of assigned bits. Codewords quantized in ADPCM quantizers 130a to 130d are multiplexed in multiplexer 150 to be a bit stream that is a multiplexed signal.
  • FIG.4 illustrates an example of quantizing bit number assignment.
  • bits shown by oblique line indicate core bits in each band.
  • the number of the core bits is five in the first band, four in the second band, three in the third band and two in the fourth band.
  • the core bits are always constant in every band, and bits assigned adaptively by adaptive bit assigner 140 are two bits shown by white in FIG.4. The two bits are assigned adaptively to each band corresponding to the energy of the dequantized value.
  • a speech decoding apparatus according to the first embodiment will be described below.
  • FIG.5 is a block diagram illustrating a configuration of the speech decoding apparatus according to the first embodiment of the present invention.
  • demultiplexer 200 decomposes an input bit stream every a number of bits assigned by adaptive bit assigner 220 described later and thus splits the bit stream into codewords for each sub-band.
  • Each of ADPCM dequantizers 210a to 210d outputs a sum of a decoded residual signal obtained by dequantizing a respective codeword and a prediction value calculated from a codeword of a last frame as a decoded sub-band signal.
  • each of ADPCM dequantizers 210a to 210d calculates a dequantized value of only core bits obtained by deleting LSB from the codeword, and the scale factor. Based on the energy of the dequantized value of the core bits calculated in each of ADPCM dequantizers 210a to 210d, adaptive bit assigner 220 calculates the number of quantizing bits assigned to the respective residual signal in the speech coding apparatus.
  • Synthesis filter bank 230 combines decoded sub-band signals output from ADPCM dequantizers 210a to 210d to obtain a decoded signal. Upsamplers 240a to 240d in synthesis filter bank 230 perform interpolation of thinned respective decoded sub-band signals. Band synthesis FIR filters 250a to 250d in synthesis filter bank 230 perform synthesis filtering on respective interpolated decoded sub-band signals. Synthesis filter bank 230 is a cosine modulation filter bank, and impulse responses of band synthesis FIR filters 250a to 250d that are basic filters are asymmetric.
  • FIG.6 is a block diagram illustrating a primary configuration of the speech decoding apparatus according to the first embodiment of the present invention. While FIG.6 illustrates a configuration of ADPCM dequantizer 210a and adaptive bit assigner 220, the other ADPCM dequantizers, 210b to 210d, have the same configuration as that of the dequantizer 210a, and are connected to adaptive bit assigner 220.
  • core bit extracting section 211 deletes LSB from the codeword input to respective one of ADPCM dequantizers 210a to 210d to extract core bits.
  • Dequantizing section 212 dequantizes the extracted core bits, and outputs a dequantized value to adder 214, predicting section 215, and adaptive bit assigner 220.
  • Scale factor adapting section 213 calculates a scale factor from the extracted core bits.
  • Adder 214 calculates the sum of the dequantized value and the prediction value calculated in predicting section 215.
  • Predicting section 215 performs zero prediction and pole prediction using the dequantized value and an output of the prediction section 215, and calculates a prediction value of a next frame of the decoded sub-band signal.
  • Dequantizing section 216 dequantizes the input codeword every a number of quantizing bits calculated in adaptive bit assigner 220 using the scale factor, and outputs a decoded residual signal.
  • Adder 217 calculates the sum of the decoded residual signal output from dequantizing section 216 and the prediction value to generate a decoded sub-band signal.
  • a bit stream input to the speech decoding apparatus is decomposed per a number of quantizing bits assigned by bit assigner 220, and thus split into codewords every four sub-bands.
  • the split codewords are input to respective ADPCM dequantizers 210a to 210d.
  • the codeword input to each of the ADPCM dequantizers 210a to 210d is dequantized in dequantizing section 216 corresponding to the number of quantizing bits assigned by adaptive bit assigner 220 and output as a decoded residual signal.
  • LSB is deleted and core bits are extracted in core bit extracting section 211.
  • the extracted core bits are input to scale factor adapting section 213 to be used in calculating a scale factor, and also to dequantizing section 212.
  • dequantizing section 212 the core bits are dequantized using the scale factor calculated in scale factor adapting section 213.
  • the dequantized value obtained by dequantizing the core bits is input to predicting section 215.
  • This input value is called a zero prediction input value.
  • the dequantized value is added in adder 214 to a prediction value of a last frame output from predicting section 215, and is input again to predicting section 215.
  • This input value is called a pole prediction input value.
  • predicting section 215 uses the zero prediction input value and pole prediction input value, predicting section 215 calculates a prediction value of a next frame of the decoded sub-band signal.
  • the dequantized value is input to adaptive bit assigner 220 per a predetermined number of frames such as a pitch period basis.
  • Adaptive bit assigner 220 calculates an energy of the dequantized value, i.e., square sum of the dequantized value as a sample, output from the each of ADPCM dequantizers 210a to 210d, and based on the calculated energy of the dequantized value, calculates the number of quantizing bits assigned to each residual signal quantized in respective one of ADPCM quantizers 130a to 130d in the speech coding apparatus.
  • the calculated numbers of quantizing bits are output to dequantizing section 216 in respective one of ADPCM dequantizers 210a to 210d, and as described above, dequantizing section 216 dequantizes a codeword of a next frame using the scale factor corresponding to the number of bits assigned in adaptive bit assigner 220 and outputs a decoded residual signal.
  • the output decoded residual signal is added in adder 217 to the prediction value output from predicting section 215 to be a decoded sub-band signal, and the decoded sub-band signal is output from each of ADPCM dequantizers 210a to 210d.
  • the decoded sub-band signals dequantized in ADPCM dequantizers 210a to 210d are subjected to interpolation in upsamplers 240a to 240d in synthesis filter bank 230, and to synthesis filtering in band synthesis FIR filters 250a to 250d.
  • the respective outputs from band synthesis FIR filters 250a to 250d are added in adders 260a to 260c to be a decoded signal.
  • synthesis filter bank 230 is a cosine modulation filter bank and impulse responses of band synthesis FIR filters 250a to 250d that are basic filters are asymmetric, a group delay occurring in filtering is decreased, and it is thereby possible to reduce an amount of computation.
  • a residual signal between a sub-band signal for each frequency band and a prediction value is quantized to output to a codeword
  • the output codeword is dequantized to calculate an energy of the dequantized value
  • the number of quantizing bits assigned in quantizing a next frame of each residual signal is determined based on the calculated energy.
  • the same codeword as that dequantized in the speech coding apparatus is dequantized to calculate the energy of the dequantized value, and based on the calculated energy, the number of quantizing bits is calculated which is determined in the speech coding apparatus to assign to a next frame of each residual signal.
  • the speech coding apparatus is capable of assigning the number of quantizing bits adaptively to each residual signal, and even when the speech coding apparatus changes the number of assigned quantizing bits, the speech decoding apparatus is capable of performing dequantization in sync with changes in the bit assignment in the speech coding apparatus without obtaining information of the changed bit assignment. Accordingly, since the speech coding apparatus does not need to notify the speech decoding apparatus of the information of the changed bit assignment to synchronize, it is possible to improve the audio quality without degrading the transmission efficiency of speech information.
  • configurations of the speech coding apparatus and speech decoding apparatus according to the second embodiment are the same as those of the speech coding apparatus and speech decoding apparatus illustrated in FIGs.2 and 5 of the first embodiment, respectively, and descriptions thereof are omitted.
  • FIG.7 is a block diagram illustrating a primary configuration of the speech coding apparatus according to the second embodiment of the present invention. While FIG.7 illustrates a configuration of ADPCM quantizer 130a and adaptive bit assigner 140a, the other ADPCM quantizers, 130b to 130d, have the same configuration as that of the quantizer 130a, and are connected to adaptive bit assigner 140a. Further, the same sections as in FIG.3 are assigned the same reference numerals to omit descriptions thereof.
  • scale factor adapting section 134a calculates a scale factor from the core bits extracted in core bit extracting section 133 to output to adaptive bit assigner 140a.
  • Dequantizing section 135a dequantizes the core bits extracted in core bit extracting section 133, and outputs a dequantized value to predicting section 136 and adder 137.
  • Adaptive bit assigner 140a determines the number of quantizing bits to assign to each of residual signals based on a scale factor calculated in respective one of ADPCM quantizers 130a to 130d.
  • Sub-band signals split in splitting filter bank 100 are input to ADPCM quantizers 130a to 130d respectively.
  • Adder 131 calculates a residual signal between the sub-band signal input to respective one of the ADPCM quantizers 130a to 130d and a prediction value of a last frame calculated in predicting section 136, and inputs the calculated residual signal to quantizing section 132.
  • the residual signal is quantized in quantizing section 132 to be a codeword with the number of quantizing bits assigned by adaptive bit assigner 140a.
  • Quantizing the residual signal uses the scale factor calculated in scale factor adapting section 134a.
  • the codeword quantized in quantizing section 132 is output to multiplexer 150, and also to core bit extracting section 133.
  • the section 133 deletes LSB to extract core bits.
  • the extracted core bits are input to scale factor adapting section 134a to be used in calculating a scale factor, and also to dequantizing section 135a.
  • the codeword quantized in quantizing section 132 becomes scalable
  • Dequantizing section 135a dequantizes the core bits using the scale factor calculated in scale factor adapting section 134a. From the dequantized value obtained by dequantizing the core bits, predicting section 136 calculates a prediction value of a next frame of the sub-band signal.
  • the scale factor is input to adaptive bit assigner 140a per a predetermined number of frames such as a pitch period basis.
  • Adaptive bit assigner 140a considers as an energy an average value of scale factors output from of ADPCM quantizers 130a to 130d, and as in the first embodiment, determines the number of quantizing bits assigned to each residual signal to be quantized in respective one of ADPCM quantizers 130a to 130d.
  • the determined numbers of quantizing bits are output to respective quantizing sections 132 in ADPCM quantizers 130a to 130d. As described above, each quantizing section 132 quantizes the residual signal of the next frame using the scale factor, and outputs a codeword with the number of assigned bits. Codewords quantized in ADPCM quantizers 130a to 130d are multiplexed in multiplexer 150 to be a bit stream that is a multiplexed signal.
  • a configuration of the speech decoding apparatus according to the second embodiment is the same as that of the speech decoding apparatus illustrated in FIG.5 of the first embodiment, and descriptions thereof are omitted.
  • FIG.8 is a block diagram illustrating a primary configuration of the speech decoding apparatus according to the second embodiment of the present invention. While FIG.8 illustrates a configuration of ADPCM dequantizer 210a and adaptive bit assigner 220a, the other ADPCM dequantizers, 210b to 210d, have the same configuration as that of the dequantizer 210a, and are connected to adaptive bit assigner 220a.
  • core bit extracting section 211 deletes LSB from the codeword input to respective one of ADPCM dequantizers 210a to 210d to extract core bits.
  • Dequantizing section 212a dequantizes the extracted core bits, and outputs a dequantized value to adder 214 and predicting section 215.
  • Scale factor adapting section 213a calculates a scale factor from the extracted core bits to output to adaptive bit assigner 220a.
  • Adder 214 calculates the sum of the dequantized value and the prediction value calculated in predicting section 215.
  • Predicting section 215 performs zero prediction and pole prediction using the dequantized value and an output of the prediction section 215, and calculates a prediction value of a next frame of the decoded sub-band signal.
  • Dequantizing section 216 dequantizes the input codeword every a number of quantizing bits calculated in adaptive bit assigner 220a using the scale factor, and outputs a decoded residual signal.
  • Adder 217 calculates the sum of the decoded residual signal output from dequantizing section 216 and the prediction value to generate a decoded sub-band signal.
  • Adaptive bit assigner 220a determines the number of quantizing bits to assign to each of residual signals based on a scale factor calculated in respective one of ADPCM dequantizers 210a to 210d.
  • Codewords split in demultiplexer 200 are input to respective ADPCM dequantizers 210a to 210d.
  • the codeword input to each of ADPCM dequantizers 210a to 210d is dequantized in dequantizing section 216 corresponding to the number of quantizing bits assigned by adaptive bit assigner 220a, and a decoded residual signal is output.
  • From the codeword input to respective one of ADPCM dequantizers 210a to 210d LSB is deleted and core bits are extracted in core bit extracting section 211.
  • the extracted core bits are input to scale factor adapting section 213a to be used in calculating a scale factor, and also to dequantizing section 212a.
  • dequantizing section 212a the core bits are dequantized using the scale factor calculated in scale factor adapting section 213a.
  • the dequantized value obtained by dequantizing the core bits is input to predicting section 215.
  • Predicting section 215 calculates a prediction value of a next frame of the decoded sub-band signal using the input dequantized value.
  • the scale factor is input to adaptive bit assigner 220a per a predetermined number of frames such as a pitch period basis.
  • Adaptive bit assigner 220a considers as an energy an average value of scale factors output from of ADPCM dequantizers 210a to 210d, and as in the first embodiment, calculates the number of quantizing bits assigned to each residual signal quantized in respective one of ADPCM quantizers 130a to 130d.
  • the calculated numbers of quantizing bits are output to dequantizing section 216 in respective one of ADPCM dequantizers 210a to 210d, and as described above, dequantizing section 216 dequantizes a codeword of a next frame using the scale factor corresponding to the number of bits assigned in adaptive bit assigner 220a and outputs a decoded residual signal.
  • the output decoded residual signal is added in adder 217 to the prediction value output from predicting section 215 to be a decoded sub-band signal, and the decoded sub-band signal is output from each of ADPCM dequantizers 210a to 210d.
  • the decoded sub-band signals dequantized in respective ADPCM dequantizers 210a to 210d are subjected to synthesis in synthesis filter bank 230 to be a decoded signal.
  • a residual signal between a sub-band signal for each frequency band and a prediction value is quantized to output a codeword
  • a scale factor is calculated from core bits of the output codeword, and based on the calculated scale factor, the number of quantizing bits assigned in quantizing a next frame of each residual signal is determined.
  • the scale factor is calculated using the same codeword as that dequantized in the speech coding apparatus, and based on the calculated scale factor, the number of quantizing bits is calculated which is determined in the speech coding apparatus to assign to a next frame of each residual signal.
  • the speech coding apparatus is capable of assigning the number of quantizing bits adaptively to each residual signal, and even when the speech coding apparatus changes the number of assigned quantizing bits, the speech decoding apparatus is capable of performing dequantization in sync with changes in the bit assignment in the speech coding apparatus without obtaining information of the changed bit assignment. Accordingly, it is possible to improve the audio quality without degrading the transmission efficiency of speech information.
  • each of the above-mentioned embodiments describes the case where an input signal is split into four sub-band signals in a splitting filter bank
  • the present invention is not limited to such a case, and it is only required to split an input signal into more than two signals corresponding to frequency band.
  • increasing the number of splits provides smoothing on signals to be quantized, and improves the following characteristic of scale factor.
  • a splitting filter bank is a cosine modulation filter
  • increasing the number of splits increases the number of taps of basic filter and suppress increases in delay amount.

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Claims (19)

  1. Dispositif de codage de la parole configuré pour exécuter un codage sur des signaux vocaux codés avec un principe à modulation ADPCM à sous-bande, comprenant :
    une section de quantification (132) configurée pour quantifier un signal résiduel d'un signal de sous-bande donné conformément à un nombre de bits affectés pour générer un mot de code, et
    une section de détermination configurée pour déterminer une valeur optimale du nombre des bits affectés utilisés dans la section de quantification (132),
    caractérisé
    par une section d'extraction (133) configurée pour extraire des bits principaux à partir du mot de code généré, et
    en ce que la section de détermination est en outre configurée pour déterminer la valeur optimale du nombre des bits affectés, affectés lors de la quantification d'une trame suivante du signal résiduel, sur la base d'une énergie correspondant aux bits principaux extraits.
  2. Dispositif de codage de la parole selon la revendication 1, dans lequel
    la section de détermination comprend une section de déquantification (135) configurée pour déquantifier les bits principaux extraits, et
    la section de détermination est en outre configurée pour déterminer, sur la base d'une énergie du signal déquantifié fourni en sortie à partir de la section de déquantification (135), une valeur optimale du nombre de bits affectés.
  3. Dispositif de codage de la parole selon la revendication 2, dans lequel la section de détermination est en outre configurée pour déterminer pour chaque période de hauteur du signal déquantifié fourni en sortie à partir de la section de déquantification (135), une valeur optimale du nombre des bits affectés sur la base de l'énergie du signal déquantifié.
  4. Dispositif de codage de la parole selon la revendication 1, dans lequel
    la section de détermination comprend une section d'acquisition de facteur d'échelle (134) configurée pour acquérir un facteur d'échelle à partir des bits principaux extraits, et
    la section de détermination est en outre configurée pour déterminer, sur la base du facteur d'échelle acquis qui est considéré comme l'énergie correspondant aux bits principaux extraits, une valeur optimale du nombre des bits affectés.
  5. Dispositif de codage de la parole selon la revendication 4, dans lequel
    la section de détermination comprend en outre une section de déquantification (135) configurée pour déquantifier les bits principaux extraits, et
    la section de détermination est en outre configurée pour déterminer pour chaque période de hauteur du signal déquantifié fourni en sortie à partir de la section de déquantification (135), une valeur optimale du nombre de bits affectés.
  6. Dispositif de codage de la parole selon la revendication 1, dans lequel la section de quantification (132) est en outre configurée pour générer les mots de code pouvant être mis à l'échelle.
  7. Dispositif de codage de la parole selon la revendication 1, comprenant en outre :
    une section de séparation (100) configurée pour séparer un signal d'entrée en une pluralité de signaux avec différentes bandes de fréquences pour générer le signal de sous-bande,
    où la section de séparation (100) comporte une batterie de filtres à modulation en cosinus, et la batterie de filtres à modulation en cosinus présente un filtre de base de sorte que sa réponse impulsionnelle soit asymétrique.
  8. Dispositif de décodage de la parole configuré pour exécuter un décodage sur des signaux vocaux codés avec un principe à modulation ADPCM à sous-bande, comprenant :
    une première section de déquantification (216) configurée pour déquantifier un mot de code donné conformément à un nombre de bits effectués pour générer un signal résiduel décodé d'un signal de sous-bande,
    une section de détermination configurée pour déterminer une valeur optimale du nombre des bits affectés utilisés dans la première section de déquantification (216),
    caractérisé
    par une section d'extraction (211) configurée pour extraire des bits principaux à partir du mot de code donné, et
    en ce que la section de détermination est en outre configurée pour déterminer la valeur optimale du nombre des bits affectés, affectés à une trame suivante du signal résiduel décodé, sur la base d'une énergie correspondant aux bits principaux extraits.
  9. Dispositif de décodage de la parole selon la revendication 8, dans lequel
    la section de détermination comprend une seconde section de déquantification (212) configurée pour déquantifier les bits principaux extraits, et
    la section de détermination est en outre configurée pour déterminer, sur la base d'une énergie du signal déquantifié fourni en sortie à partir de la seconde section de déquantification (212), une valeur optimale du nombre des bits affectés.
  10. Dispositif de décodage de la parole selon la revendication 9, dans lequel la section de détermination est en outre configurée pour déterminer pour chaque période de hauteur du signal déquantifié fourni en sortie à partir de la seconde section de déquantification (212), une valeur optimale du nombre des bits affectés.
  11. Dispositif de décodage de la parole selon la revendication 8, dans lequel
    la section de détermination comprend une section d'acquisition de facteur d'échelle (213) configurée pour acquérir un facteur d'échelle à partir des bits principaux extraits, et
    la section de détermination est en outre configurée pour déterminer, sur la base du facteur d'échelle acquis qui est considéré comme l'énergie correspondant aux bits principaux extraits, une valeur optimale du nombre des bits affectés.
  12. Dispositif de décodage de la parole selon la revendication 11, dans lequel
    la section de détermination comprend en outre une seconde section de déquantification (212) configurée pour déquantifier les bits principaux extraits, et
    la section de détermination est en outre configurée pour déterminer pour chaque période de hauteur du signal déquantifié fourni en sortie à partir de la seconde section de déquantification (212), une valeur optimale du nombre des bits affectés.
  13. Dispositif de décodage de la parole selon la revendication 8, comprenant en outre :
    une section de synthèse (230) configurée pour exécuter une synthèse sur les signaux de sous-bande décodés générés dans les sections de génération (210a, 210b, 210c, 210d)
    où la section de synthèse (230) comporte une batterie de filtres à modulation en cosinus, et la batterie de filtres à modulation en cosinus comporte un filtre de base de sorte que sa réponse impulsionnelle soit asymétrique.
  14. Système de transmission de microphone sans fil numérique comportant le dispositif de codage de la parole selon la revendication 1.
  15. Système de réception de microphone sans fil numérique comportant un dispositif de décodage de la parole selon la revendication 8.
  16. Procédé de codage de la parole consistant à exécuter un codage sur des signaux vocaux codés avec un principe à modulation ADPCM à sous-bande, comprenant :
    une étape de quantification consistant à quantifier un signal résiduel d'un signal de sous-bande donné conformément à un nombre de bits affectés pour générer un mot de code,
    une étape d'acquisition consistant à acquérir une valeur optimale du nombre des bits affectés, et
    une étape de quantification consistant à quantifier un signal résiduel d'une trame suivante d'un signal de sous-bande donné conformément à la valeur optimale acquise du nombre des bits affectés pour générer un mot de code de la trame suivante,
    caractérisé
    par une étape d'extraction consistant à extraire des bits principaux à partir du mot de code généré, et
    en ce que l'étape d'acquisition acquiert la valeur optimale du nombre des bits affectés, affectés lors de la quantification de la trame suivante du signal résiduel, sur la base d'une énergie correspondant aux bits principaux extraits.
  17. Procédé de décodage de la parole consistant à exécuter un décodage sur des signaux vocaux codés avec un principe à modulation ADPCM à sous-bande, comprenant :
    une étape de déquantification consistant à déquantifier un mot de code donné conformément à un nombre de bits affectés pour générer un signal résiduel décodé d'un signal de sous-bande,
    une étape d'acquisition consistant à acquérir une valeur optimale du nombre des bits affectés, et
    une étape de déquantification consistant à déquantifier un mot de code d'une trame suivante conformément à la valeur optimale acquise du nombre des bits affectés pour générer un signal résiduel décodé de la trame suivante d'un signal de sous-bande,
    caractérisé
    par une étape d'extraction consistant à extraire des bits principaux à partir du mot de code donné, et
    en ce que l'étape d'acquisition acquiert la valeur optimale du nombre des bits affectés, affectés à la trame suivante du signal résiduel décodé sur la base d'une énergie correspondant aux bits principaux extraits.
  18. Procédé de décodage de la parole selon la revendication 17, dans lequel dans l'étape d'acquisition, le même mot de code est déquantifié, lequel a été utilisé pour déterminer une valeur optimale du nombre des bits affectés durant le codage, et sur la base d'une énergie d'un signal déquantifié de sortie, une valeur optimale du nombre des bits affectés est acquise.
  19. Procédé de décodage de la parole selon la revendication 17, dans lequel dans l'étape d'acquisition, les mêmes bits principaux que ceux d'un mot de code utilisé durant le codage sont extraits, un facteur d'échelle est calculé à partir des bits principaux extraits et sur la base du facteur d'échelle calculé, une valeur optimale du nombre des bits affectés est déterminée.
EP02025094A 2001-11-13 2002-11-12 Procédé et dispositif de codage et de décodage de signaux de parole Expired - Lifetime EP1310943B1 (fr)

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JP2001347408 2001-11-13
JP2001347408A JP4245288B2 (ja) 2001-11-13 2001-11-13 音声符号化装置および音声復号化装置

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EP1310943B1 true EP1310943B1 (fr) 2007-01-17

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US7155384B2 (en) 2006-12-26
CN1419349A (zh) 2003-05-21
JP2003150198A (ja) 2003-05-23
US20030093266A1 (en) 2003-05-15
DE60217612T2 (de) 2007-05-16
EP1310943A2 (fr) 2003-05-14
CN100440758C (zh) 2008-12-03
EP1310943A3 (fr) 2004-02-11
JP4245288B2 (ja) 2009-03-25
DE60217612D1 (de) 2007-03-08

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